In this description, the term sending communication device refers to a communication device including a transmitter being arranged to send multimedia streams to a communication network. The term receiving communication device refers to a communication device including a receiver for receiving multimedia streams from the communication network, respectively. It is obvious that the same communication device may include both the transmitter and the receiver whereby allowing one-way or two-way communication with the communication network. A wireless communication device includes a transmitter and/or a receiver implementing wireless communication in a wireless communication network. The term wireless communication system, such as a mobile communication system, generally refers to any communication system which makes a wireless data transmission connection possible between a wireless communication device and stationary parts of the system, the user of the wireless communication device moving within the operating range of the system. A typical wireless communication system is a public land mobile network PLMN.
A well-known example is the GSM system (Global System for Mobile Telecommunications). The invention preferably relates to the third generation of mobile communication systems. As an example, the Universal Mobile Telecommunications System UMTS is used as an example of such a third-generation communication system.
In third generation systems, the terms bearer service and service are used. A bearer service is a telecommunication service type which provides the facility to transmit signals between access points. In general, the bearer service corresponds to the older term of a traffic channel which defines, for example, the data transmission rate and the quality of service (QoS) to be used in the system when information is transmitted between a wireless communication device and another part of the system. The bearer service between the wireless communication device and the base station is, for example, a radio bearer service, and the bearer service between the radio network control unit and the core network is, for example, an Iu bearer service (Interface UMTS bearer). In the UMTS system, the interface between the radio network control unit and the core network is called Iu interface. In UMTS there is also the so called GERAN part, which uses, in addition to the Iu interface, also an interface called as Gb interface. In this connection, the service is provided by the mobile communication network for performing a task (tasks); for example, data services perform data transmission in the communication system, telephone services are related to telephone calls, multimedia, etc. Thus, the service requires data transmission, such as a telephone call or the transmission of multimedia streams, between the wireless communication device and the stationary parts of the system. One important task of the operation of a third-generation mobile communication system is to control (initialize, maintain and terminate, according to the need) bearer services in such a way that each requested service can be allocated to mobile stations without wasting the available bandwidth.
The quality of service determines, for example, how protocol data units (PDU) are processed in the mobile communication network during the transmission. For example, QoS levels defined for connection addresses are used for controlling the transmission order, buffering (packet strings) and rejecting packets in support nodes and gateway support nodes, particularly when two or more connections have packets to be transmitted simultaneously. The different QoS levels determine, for example, different delays for packet transmissions between the different ends of the connection, as well as different bit rates. Also, the number of rejected and/or lost packet data units may vary in connections with different QoS levels.
It is possible to request for a different QoS for each PDP context. For example, in e-mail connections, a relatively long delay can be allowed in the transmission of streams. However, real-time interactive applications, such as video conferencing, require packet transmission at a high rate. In some applications, such as file transfers, it is important that the packet switched transmission is faultless, wherein in error situations, the packet data units are retransmitted, if necessary.
For the packet switched communication service in the UMTS system, the defining of four different traffic classes has been proposed, and for the properties of these traffic classes, the aim has been to consider the different criteria for the different connection types. One criterion defined for the first and second classes is that the transmission takes place in real time, wherein the transmission must have no significant delays. However, in such classes, the accuracy of the data transfer is not such an important property. In a corresponding manner, non-real time data transmission is sufficient for the third and fourth traffic classes, but a relatively accurate data transmission is required of them. An example of real-time first-class communication is the transmission of speech signals in a situation in which two or more persons are discussing with each other by means of wireless communication devices. An example of a situation in which real-time second-class communication might be feasible, is the transmission of a video signal for immediate viewing. Third-class non-real time packet communication can be used, for example, for the use of database services, such as the browsing of Internet home pages, in which the relatively accurate data transmission at a reasonable rate is a more important factor than the real-time data transmission. In the system according to this example, for example the transfer of e-mail messages and files can be classified to the fourth category. Naturally, the number of traffic classes is not necessarily four as mentioned here, but the invention can be applied in packet switched communication systems comprising any number of traffic classes. The properties of the four presented traffic classes are briefly presented in Table 1.
TABLE 1Second class (stream-Third class (inter-Fourth classing class):active class):(background class):real-time, e.g. videointeractive bestbackground trans-First class (conversa-informationeffort methodmission by the besttional class):guaranteed capacityacknowledgementeffort methodreal-time, e.g. telephoneacknowledgementInternet browser,acknowledgementconversationpossibleTelnetbackground loadingguaranteed capacitybuffering on applica-real-time controlof e-mail messages,Classno acknowledgementtion levelchannelcalendar events, etc.Maximum<2048<2048<2048 − overhead<2048 − overheadbit rateDeliveryYes/NoYes/NoYes/NoYes/NoorderMaximum≦1500 or 1502≦1500 or 1502≦1500 or 1502≦1500 or 1502packet size(SDU)Transmis-Yes/No/−Yes/No/−Yes/No/−Yes/No/−sion ofincorrectpackets(SDU)Residual5*10−2, 10−2, 5*10−3, 10−3,5*10−2, 10−2, 5*10−3,4*10−3, 10−5, 6*10−84*10−3, 10−5, 6*10−8bit error10−4, 10−5, 10−610−3, 10−4, 10−5, 10−6ratioPacket10−2, 7*10−3, 10−3, 10−4,10−1, 10−2, 7*10−3,10−3, 10−4, 10−610−3, 10−4, 10−6error ratio10−510−3, 10−4, 10−5(SDU)Trans-100 ms—maximum250 ms—maximum missionvaluevaluedelayGuaranteed<2048<2048bit rateTraffic1, 2, 3processingpriorityAllocation1, 2, 31, 2, 31, 2, 31, 2, 3priority
The guaranteed bit rate is used for admission control and resource reservation at the RAN and CN, the maximum bit rate is used for policing at the CN, i.e. no higher than the maximum bit rate is allowed to enter the CN at the GGSN, packets that exceed this bit rate will be dropped.
Modern second and third generation wireless communication devices have much better data processing properties than older wireless communication devices. For example, they already have the facility of connecting to the Internet and using a browsing application in the wireless communication device to retrieve information from the Internet, and in the future, it will be possible to set up multimedia calls, for example, for real-time video conferences and the like.
The requirements of different applications may be significantly different. Some applications require fast communication between the sender and the receiver. These applications include, for example, video and telephone applications. Some other applications may require as accurate data transmission as possible, but the bit rate of the data transmission connection is less important. These applications include, for example, e-mail and database applications. On the other hand, these applications can be used in several wireless communication devices with different properties.
The user of the wireless communication device may be willing to watch a multimedia presentation with the wireless communication device. The user finds the loading address of such a presentation and sends a request to send the presentation to the wireless communication device. The request is handled in the communication system. The loading address of the requested multimedia presentation may address to a server in a communication network, such as a server of the Internet. The server which delivers the multimedia presentation to the receiving wireless communication device is called as a streaming server in this description.
The communication system should reserve enough resources for the communication between the streaming server and the wireless communication device to be able to deliver the requested multimedia presentation. Otherwise the presentation may not be presented with the same accuracy and error free in the receiving wireless communication device. In the UMTS communication system the wireless communication device requests a PDP context with certain QoS parameters first. Then, the network selects a bearer service for the connection by using some selection bases, for example, the parameters the wireless communication device has possible used in the request. Such selection bases may not be appropriate or accurate enough wherein situations may occur in which the bearer service can not provide enough transmission capacity for the connection, or it provides more capacity than is needed, wherein the usage of the network resources is not efficient.
Another situation in which a delivery of multimedia information may be needed is two wireless communication devices communicating with each other to exchange multimedia information such as video or still images. Also in this kind of situation enough resources should be reserved by the network for the communication. However, when using prior art methods it is not always possible to inform both ends of the connection about the demands for the connection.
Currently there is no way to signal end-to-end what is the maximum bit rate used by an application, for example, a video application. A streaming server is not able to signal the maximum bit rate of the encoded multimedia stream to the streaming client (the receiving wireless communication device). The latter has only information about the guaranteed bit rate, but not about the maximum bit rate. The client can then make three kinds of decisions. First, the client may choose a maximum bit rate (MBR) value that is equal to the guaranteed bit rate (GBR). This could cause packet losses and bad received quality whenever the bit rate exceeds the guaranteed bit rate (=maximum bit rate). For example, if the GBR=MBR=60 kbps and a compressed video source bit rate is encoded at 60 kbps on average, but some sporadic high bit rate peaks at 64 kbps are occurring, then the effect would be a certain period of packet losses at the receiving end (the period is equal to the time the bit rate exceeds 60 kbps). To avoid this situation, the MBR and GBR could be set in a way such that GBR=MBR=64 kbps. This would avoid packet losses, but it translates into an inefficient way of handling network resources through over-engineering, because the bandwidth between 60 and 64 kbps would not be used all the time, producing a waste of 4 kbps on average.
Second, the client may choose a maximum bit rate higher than the guaranteed bit rate by making some estimations. These estimations can be inaccurate, because even if the client uses past history information about the bit rate, the maximum bit rate used by a generic server cannot be easily predicted. Also in this case packet losses can occur.
Third, the client may choose a very high maximum bit rate value, in order to get a downgrading from the network to the maximum subscribed bit rate.
The second and third solutions would yield some inefficiencies because the streaming server would not be informed about the maximum bit rate of the UMTS bearer, making possibly wrong assumptions about transmission bit rate and bandwidth adaptation algorithms.
A conversational multimedia application in a mobile communication device is not able to signal the maximum bit rate of the session to the other mobile communication device. This means that each communication device (symmetrically) is able to know at what guaranteed bit rate the other communication device will encode the multimedia streams. However, each communication device (symmetrically) will not be able to know what is the downlink maximum bit rate. In other words, each communication device will not be able to know at what maximum bit rate the other communication device will encode the multimedia streams.
Also in this case the mobile communication device can decide to choose one of the three alternatives mentioned above. Also in this situation the choice of any of the mentioned alternatives would cause similar problems as described above for streaming, since each communication device would not know at what maximum bit rate the respective encoders will encode the media flows.
In present systems, the wireless communication device and the mobile communication network negotiate to select such a bearer service with which the QoS requirements can be fulfilled. For example, in the system according to the UMTS standard, the wireless communication device may freely request for a desired quality of service, wherein the UMTS mobile communication network examines if it can provide the quality of service requested by the wireless communication device. If the application to be executed in the wireless communication device contains QoS requirements, the wireless communication device transmits these QoS requirements as such to the mobile communication network, for the selection of the bearer service. However, if the application does not transmit QoS requirements to the wireless communication device, a default QoS profile stored in the network is normally used (typically in the Home Location Register, HLR), in which certain properties have been predetermined for the connection. If the properties of the wireless communication device do not, in all respects, meet the quality of service requested for the application, the performance of the application is probably not appropriate.
In addition to the maximum bit rate there is another parameter, the maximum service data unit (SDU) size parameter, which is not known by the other party of the connection, i.e. the streaming server and/or the sending communication device. The SDU size parameter describes the size of the packets of the multimedia stream transmitted by the streaming server. Therefore, the streaming client has to select for the maximum SDU size such a value which is big enough for the streaming client to be able receive all the packets. The streaming client may try an arbitrary value or it may select the maximum allowable value for the SDU size. This kind of selection may cause that unnecessary resources will be reserved for the multimedia streaming session.
It may also happen that when the streaming client sends a resource allocation request to the network (indicating, for example, maximum bit rate and maximum SDU size parameters in the request), the network may not (or cannot) reserve the requested resources. In prior art systems the streaming client can not inform the streaming server about the allocated resources. It may then happen that the streaming server sends the multimedia stream in larger data units than is appropriate for the connection between the wireless communication network and the wireless communication device.
If a certain fixed number of bytes (e.g. 1500 bytes) would be used all the time in the QoS profile, this would cause inefficiencies in the network and it could yield lower media quality to the packet switched streaming client. In fact, the network assuming all the packets of fixed size, would have more difficulties in maintaining the target SDU error ratio with the given delay in the QoS profile, because the larger the packets the more difficult it is to keep the target SDU error rate below a predetermined value.